22 Feb
2017
22 Feb
'17
10:02 a.m.
On 21 Feb 2017, at 21:40, Keith
<keith(a)rhizomatica.org> wrote:
Hi Keith,
On the question about SIP re-invite, I am talking
about operating
without rtp_proxy, so that one can have the advantage of BTS<->BTS RTP
streams at the same time as handover. From what I've read, this is quite
feasable, as part of the SIP spec.
I think this is already considered as part of the development of the
osmo-sip-connector, which is a project I really want to see moving
forward and hope to find time to contribute to over the next few months.
yes for the osmo-sip-connector I have ignored the topic of handover. In
general I try to see how long I get away with not having to touch the
UDP/RTP streams.
In case of hand-over the new BTS might send from a different src IP,
src Port and will most likely have a new SSRC, timestamp seqno. It is
a bit of a question of how "your" PBX will handle it. I can see a
couple of outcomes.
a.) It has some support for "NAT" handling and will just use the new
stream and return packages to the new src IP/src port. It might go back
and forth but at some point the old bts will stop sending things.
b.) It will accept the new stream but will try to send to the old BTS.
The we would have one-way audio.
c.) It will reject it as it is a unknown src ip/port.
I think for b.) and c.) we will require SIP re-invite but also need
to look at the AoIP spec to see if they say how to handle this
scenario. But that depends on how the PBX handles this as well. So it
means for handover we always need to look at BSC/SIP+PBX and how it is
handling SDP/RTP.
cheers
holger
PS: Do you have a docker container for your FreeSWITCH setup and maybe
two numbers one to play audio and another to record the audio?